Stream audio, live.

Uncompressed PCM or low-delay Opus over a WebSocket, played through the Web Audio API with an adaptive jitter buffer. No media element, no HLS, no buffering bar.

Room Sender and receiver must use the same room name.

Audio source

Capture from

Device

Device names appear after you grant microphone access once.

Disable echo cancellation, noise suppression and AGC (required for full-quality music)

Format

Codec

Channels

Frame size

Opus bitrate 128 kbps

Estimated wire rate: --

-60-30-12-60 dB

Idle.

Receiver link

Open this on any other device on the network. Over plain HTTP only localhost works — see the README for the one-command HTTPS setup.

Sender statistics

Listeners0

Uptime--

Frames sent0

Wire rate--

Sent0 B

Sample rate--

Server RTT--

Clock offset--

Capture latency--

Dropped (backpressure)0

Tap listen to connect.

Volume 100%

Jitter buffer target 60 ms

Lower is less delay, higher survives worse networks. Raise it if you hear gaps.

Route through a media element (iOS: plays even with the ringer switch on mute; adds a little delay) Keep screen awake while listening Recover automatically from lag (re-primes the buffer, and reconnects if the stream falls behind live)

Live statistics

End-to-end delay--

Network delay--

Buffer--

Jitter--

Packet loss--

Loss (last 10 s)--

Packets received0

Packets lost0

Out of order0

Underruns0

Overruns0

Lag vs. best--

Auto-resyncs0

Clock drift--

Bitrate--

Received0 B

Server RTT--

Output latency--

Stream--

Playback rate--

History delay buffer loss

Last 60 seconds.